Recent advances in LVCSR : A benchmark comparison of performances

Large Vocabulary Continuous Speech Recognition (LVCSR), which is characterized by a high variability of the speech, is the most challenging task in automatic speech recognition (ASR). Believing that the evaluation of ASR systems on relevant and common speech corpora is one of the key factors that help accelerating research, we present, in this paper, a benchmark comparison of the performances of the current state-of-the-art LVCSR systems over different speech recognition tasks. Furthermore, we put objectively into evidence the best performing technologies and the best accuracy achieved so far in each task. The benchmarks have shown that the Deep Neural Networks and Convolutional Neural Networks have proven their efficiency on several LVCSR tasks by outperforming the traditional Hidden Markov Models and Guaussian Mixture Models. They have also shown that despite the satisfying performances in some LVCSR tasks, the problem of large-vocabulary speech recognition is far from being solved in some others, where more research efforts are still needed

Word hypothesis of phonetic strings using hidden Markov models

This thesis investigates a stochastic modeling approach to word hypothesis of phonetic strings for a speaker independent, large vocabulary, continuous speech recognition system. The stochastic modeling technique used is Hidden Markov Modeling. Hidden Markov Models (HMM) are probabilistic modeling tools most often used to analyze complex systems. This thesis is part of a speaker independent, large vocabulary, continuous speech understanding system under development at the Rochester Institute of Technology Research Corporation. The system is primarily data-driven and is void of complex control structures such as the blackboard approach used in many expert systems. The software modules used to implement the HMM were created in COMMON LISP on a Texas Instruments Explorer II workstation. The HMM was initially tested on a digit lexicon and then scaled up to a U.S. Air Force cockpit lexicon. A sensitivity analysis was conducted using varying error rates. The results are discussed and a comparison with Dynamic Time Warping results is made

Word-Graph Based Applications for Handwriting Documents: Impact of Word-Graph Size on Their Performances

The final publication is available at Springer via http://dx.doi.org/10.1007/978-3-319-19390-8 29Computer Assisted Transcription of Text Images (CATTI) and Key-Word Spotting (KWS) applications aim at transcribing and indexing handwritten documents respectively. They both are approached by means of Word Graphs (WG) obtained using segmentation-free handwritten text recognition technology based on N-gram Language Models and Hidden Markov Models. A large WG contains most of the relevant information of the original text (line) image needed for CATTI and KWS but, if it is too large, the computational cost of generating and using it can become unaffordable. Conversely, if it is too small, relevant information may be lost, leading to a reduction of CATTI/KWS in performance accuracy. We study the trade-off between WG size and CATTI &KWS performance in terms of effectiveness and efficiency. Results show that small, computationally cheap WGs can be used without loosing the excellent CATTI/KWS performance achieved with huge WGs.Work partially supported by the Spanish MICINN projects STraDA (TIN2012-37475-C02-01) and by the EU 7th FP tranScriptorium project (Ref:600707).Toselli, AH.; Romero Gómez, V.; Vidal Ruiz, E. (2015). Word-Graph Based Applications for Handwriting Documents: Impact of Word-Graph Size on Their Performances. En Pattern Recognition and Image Analysis. Springer. 253-261. https://doi.org/10.1007/978-3-319-19390-8_29S253261Romero, V., Toselli, A.H., Vidal, E.: Multimodal Interactive Handwritten Text Transcription. Series in Machine Perception and Artificial Intelligence (MPAI). World Scientific Publishing, Singapore (2012)Toselli, A.H., Vidal, E., Romero, V., Frinken, V.: Word-graph based keyword spotting and indexing of handwritten document images. Technical report, Universitat Politècnica de València (2013)Oerder, M., Ney, H.: Word graphs: an efficient interface between continuous-speech recognition and language understanding. In: IEEE International Conference on Acoustics, Speech, and Signal Processing, vol. 2, pp. 119–122, April 1993Bazzi, I., Schwartz, R., Makhoul, J.: An omnifont open-vocabulary OCR system for English and Arabic. IEEE Trans. Pattern Anal. Mach. Intell. 21(6), 495–504 (1999)Jelinek, F.: Statistical Methods for Speech Recognition. MIT Press, Cambridge (1998)Ström, N.: Generation and minimization of word graphs in continuous speech recognition. In: Proceedings of IEEE Workshop on ASR 1995, Snowbird, Utah, pp. 125–126 (1995)Ortmanns, S., Ney, H., Aubert, X.: A word graph algorithm for large vocabulary continuous speech recognition. Comput. Speech Lang. 11(1), 43–72 (1997)Wessel, F., Schluter, R., Macherey, K., Ney, H.: Confidence measures for large vocabulary continuous speech recognition. IEEE Trans. Speech Audio Process. 9(3), 288–298 (2001)Robertson, S.: A new interpretation of average precision. In: Proceedings of the International ACM SIGIR Conference on Research and Development in Information Retrieval (SIGIR 2008), pp. 689–690. ACM, USA (2008)Manning, C.D., Raghavan, P., Schutze, H.: Introduction to Information Retrieval. Cambridge University Press, USA (2008)Romero, V., Toselli, A.H., Rodríguez, L., Vidal, E.: Computer assisted transcription for ancient text images. In: Kamel, M.S., Campilho, A. (eds.) ICIAR 2007. LNCS, vol. 4633, pp. 1182–1193. Springer, Heidelberg (2007)Fischer, A., Wuthrich, M., Liwicki, M., Frinken, V., Bunke, H., Viehhauser, G., Stolz, M.: Automatic transcription of handwritten medieval documents. In: 15th International Conference on Virtual Systems and Multimedia, VSMM 2009, pp. 137–142 (2009)Pesch, H., Hamdani, M., Forster, J., Ney, H.: Analysis of preprocessing techniques for latin handwriting recognition. In: ICFHR, pp. 280–284 (2012)Evermann, G.: Minimum Word Error Rate Decoding. Ph.D. thesis, Churchill College, University of Cambridge (1999

Surrogate Gradient Spiking Neural Networks as Encoders for Large Vocabulary Continuous Speech Recognition

Compared to conventional artificial neurons that produce dense and real-valued responses, biologically-inspired spiking neurons transmit sparse and binary information, which can also lead to energy-efficient implementations. Recent research has shown that spiking neural networks can be trained like standard recurrent neural networks using the surrogate gradient method. They have shown promising results on speech command recognition tasks. Using the same technique, we show that they are scalable to large vocabulary continuous speech recognition, where they are capable of replacing LSTMs in the encoder with only minor loss of performance. This suggests that they may be applicable to more involved sequence-to-sequence tasks. Moreover, in contrast to their recurrent non-spiking counterparts, they show robustness to exploding gradient problems without the need to use gates

Speed perturbation and vowel duration modeling for ASR in Hausa and Wolof languages

International audienceAutomatic Speech Recognition (ASR) for (under-resourced) Sub-Saharan African languages faces several challenges: small amount of transcribed speech, written language normalization issues, few text resources available for language modeling, as well as specific features (tones, morphology, etc.) that need to be taken into account seriously to optimize ASR performance. This paper tries to address some of the above challenges through the development of ASR systems for two Sub-Saharan African languages: Hausa and Wolof. First, we investigate data augmentation technique (through speed perturbation) to overcome the lack of resources. Secondly, the main contribution is our attempt to model vowel length contrast existing in both languages. For reproducible experiments, the ASR systems developed for Hausa and Wolof are made available to the research community on github. To our knowledge, the Wolof ASR system presented in this paper is the first large vocabulary continuous speech recognition system ever developed for this language

Segmentation, Diarization and Speech Transcription: Surprise Data Unraveled

In this thesis, research on large vocabulary continuous speech recognition for unknown audio conditions is presented. For automatic speech recognition systems based on statistical methods, it is important that the conditions of the audio used for training the statistical models match the conditions of the audio to be processed. Any mismatch will decrease the accuracy of the recognition. If it is unpredictable what kind of data can be expected, or in other words if the conditions of the audio to be processed are unknown, it is impossible to tune the models. If the material consists of `surprise data' the output of the system is likely to be poor. In this thesis methods are presented for which no external training data is required for training models. These novel methods have been implemented in a large vocabulary continuous speech recognition system called SHoUT. This system consists of three subsystems: speech/non-speech classification, speaker diarization and automatic speech recognition. The speech/non-speech classification subsystem separates speech from silence and unknown audible non-speech events. The type of non-speech present in audio recordings can vary from paper shuffling in recordings of meetings to sound effects in television shows. Because it is unknown what type of non-speech needs to be detected, it is not possible to train high quality statistical models for each type of non-speech sound. The speech/non-speech classification subsystem, also called the speech activity detection subsystem, does not attempt to classify all audible non-speech in a single run. Instead, first a bootstrap speech/silence classification is obtained using a standard speech activity component. Next, the models for speech, silence and audible non-speech are trained on the target audio using the bootstrap classification. This approach makes it possible to classify speech and non-speech with high accuracy, without the need to know what kinds of sound are present in the audio recording. Once all non-speech is filtered out of the audio, it is the task of the speaker diarization subsystem to determine how many speakers occur in the recording and exactly when they are speaking. The speaker diarization subsystem applies agglomerative clustering to create clusters of speech fragments for each speaker in the recording. First, statistical speaker models are created on random chunks of the recording and by iteratively realigning the data, retraining the models and merging models that represent the same speaker, accurate speaker models are obtained for speaker clustering. This method does not require any statistical models developed on a training set, which makes the diarization subsystem insensitive for variation in audio conditions. Unfortunately, because the algorithm is of complexity $O(n^3)$, this clustering method is slow for long recordings. Two variations of the subsystem are presented that reduce the needed computational effort, so that the subsystem is applicable for long audio recordings as well. The automatic speech recognition subsystem developed for this research, is based on Viterbi decoding on a fixed pronunciation prefix tree. Using the fixed tree, a flexible modular decoder could be developed, but it was not straightforward to apply full language model look-ahead efficiently. In this thesis a novel method is discussed that makes it possible to apply language model look-ahead effectively on the fixed tree. Also, to obtain higher speech recognition accuracy on audio with unknown acoustical conditions, a selection from the numerous known methods that exist for robust automatic speech recognition is applied and evaluated in this thesis. The three individual subsystems as well as the entire system have been successfully evaluated on three international benchmarks. The diarization subsystem has been evaluated at the NIST RT06s benchmark and the speech activity detection subsystem has been tested at RT07s. The entire system was evaluated at N-Best, the first automatic speech recognition benchmark for Dutch

Word graphs size impact on the performance of handwriting document applications

[EN] Two document processing applications are con- sidered: computer-assisted transcription of text images (CATTI) and Keyword Spotting (KWS), for transcribing and indexing handwritten documents, respectively. Instead of working directly on the handwriting images, both of them employ meta-data structures called word graphs (WG), which are obtained using segmentation-free hand- written text recognition technology based on N-gram lan- guage models and hidden Markov models. A WG contains most of the relevant information of the original text (line) image required by CATTI and KWS but, if it is too large, the computational cost of generating and using it can become unafordable. Conversely, if it is too small, relevant information may be lost, leading to a reduction of CATTI or KWS performance. We study the trade-off between WG size and performance in terms of effectiveness and effi- ciency of CATTI and KWS. Results show that small, computationally cheap WGs can be used without loosing the excellent CATTI and KWS performance achieved with huge WGs.Work partially supported by the Generalitat Valenciana under the Prometeo/2009/014 Project Grant ALMAMATER, by the Spanish MECD as part of the Valorization and I+D+I Resources program of VLC/CAMPUS in the International Excellence Campus program, and through the EU projects: HIMANIS (JPICH programme, Spanish Grant Ref. PCIN-2015-068) and READ (Horizon-2020 programme, Grant Ref. 674943).Toselli ., AH.; Romero Gómez, V.; Vidal, E. (2017). Word graphs size impact on the performance of handwriting document applications. Neural Computing and Applications. 28(9):2477-2487. https://doi.org/10.1007/s00521-016-2336-2S24772487289Amengual JC, Vidal E (1998) Efficient error-correcting Viterbi parsing. IEEE Trans Pattern Anal Mach Intell 20(10):1109–1116Bazzi I, Schwartz R, Makhoul J (1999) An omnifont open-vocabulary OCR system for English and Arabic. 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The TransLectures-UPV Toolkit

The final publication is available at Springer via http://dx.doi.org/10.1007/978-3-319-13623-3_28Over the past few years, online multimedia educational repositories have increased in number and popularity. The main aim of the transLectures project is to develop cost-effective solutions for producing accurate transcriptions and translations for large video lecture repositories, such as VideoLectures.NET or the Universitat Politècnica de València s repository, poliMedia. In this paper, we present the transLectures-UPV toolkit (TLK), which has been specifically designed to meet the requirements of the transLectures project, but can also be used as a conventional ASR toolkit. The main features of the current release include HMM training and decoding with speaker adaptation techniques (fCMLLR). TLK has been tested on the VideoLectures.NET and poliMedia repositories, yielding very competitive results. TLK has been released under the permissive open source Apache License v2.0 and can be directly downloaded from the transLectures website.The research leading to these results has received funding from the European Union Seventh Framework Programme (FP7/2007-2013) under grant agreement no 287755 (transLectures) and ICT Policy Support Programme (ICT PSP/2007-2013) as part of the Competitiveness and InnovationFramework Programme (CIP) under grant agreement no 621030 (EMMA), andthe Spanish MINECO Active2Trans (TIN2012-31723) research project.Del Agua Teba, MA.; Giménez Pastor, A.; Serrano Martinez Santos, N.; Andrés Ferrer, J.; Civera Saiz, J.; Sanchis Navarro, JA.; Juan Císcar, A. (2014). The TransLectures-UPV Toolkit. En Advances in Speech and Language Technologies for Iberian Languages: Second International Conference, IberSPEECH 2014, Las Palmas de Gran Canaria, Spain, November 19-21, 2014. Proceedings. Springer International Publishing. 269-278. https://doi.org/10.1007/978-3-319-13623-3_28S269278Final report on massive adaptation (M36). To be delivered on October 2014 (2014)First report on massive adaptation (M12), https://www.translectures.eu/wp-content/uploads/2013/05/transLectures-D3.1.1-18Nov2012.pdfOpencast Matterhorn, http://opencast.org/matterhorn/sclite - Score speech recognition system output, http://www1.icsi.berkeley.edu/Speech/docs/sctk-1.2/sclite.htmSecond report on massive adaptation (M24), https://www.translectures.eu//wp-content/uploads/2014/01/transLectures-D3.1.2-15Nov2013.pdfTLK: The transLectures-UPV Toolkit, https://www.translectures.eu/tlk/Baum, L.E., Petrie, T., Soules, G., Weiss, N.: A Maximization Technique Occurring in the Statistical Analysis of Probabilistic Functions of Markov Chains. The Annals of Mathematical Statistics 41(1), 164–171 (1970)Dahl, G.E., Yu, D., Deng, L., Acero, A.: Context-Dependent Pre-Trained Deep Neural Networks for Large-Vocabulary Speech Recognition. IEEE Transactions on Audio, Speech, and Language Processing 20(1), 30–42 (2012)Digalakis, V., Rtischev, D., Neumeyer, L., Sa, E.: Speaker Adaptation Using Constrained Estimation of Gaussian Mixtures. IEEE Transactions on Speech and Audio Processing 3, 357–366 (1995)Huang, J.T., Li, J., Yu, D., Deng, L., Gong, Y.: Cross-language knowledge transfer using multilingual deep neural network with shared hidden layers. In: Proc. of ICASSP (2013)Munteanu, C., Baecker, R., Penn, G., Toms, E., James, D.: The Effect of Speech Recognition Accuracy Rates on the Usefulness and Usability of Webcast Archives. In: Proc. of CHI, pp. 493–502 (2006)Ney, H., Ortmanns, S.: Progress in dynamic programming search for LVCSR. Proceedings of the IEEE 88(8), 1224–1240 (2000)Ortmanns, S., Ney, H., Eiden, A.: Language-model look-ahead for large vocabulary speech recognition. In: Proc. of ICSLP, vol. 4, pp. 2095–2098 (1996)Ortmanns, S., Ney, H., Aubert, X.: A word graph algorithm for large vocabulary continuous speech recognition. Computer Speech and Language 11(1), 43–72 (1997)Povey, D., et al.: The Kaldi Speech Recognition Toolkit. In: Proc. of ASRU (2011)Rumelhart, D., Hintont, G., Williams, R.: Learning representations by back-propagating errors. Nature 323(6088), 533–536 (1986)Rybach, D., et al.: The RWTH Aachen University Open Source Speech Recognition System. In: Proc. Interspeech, pp. 2111–2114 (2009)Seide, F., Li, G., Chen, X., Yu, D.: Feature engineering in Context-Dependent Deep Neural Networks for conversational speech transcription. In: Proc. of ASRU, pp. 24–29 (2011)Viterbi, A.: Error bounds for convolutional codes and an asymptotically optimum decoding algorithm. IEEE Transactions on Information Theory 13(2), 260–269 (1967)Young, S., et al.: The HTK Book. Cambridge University Engineering Department (1995)Young, S.J., Odell, J.J., Woodland, P.C.: Tree-based state tying for high accuracy acoustic modelling. In: Proc. of HLT, pp. 307–312 (1994